| 00:00.10 | leifmadsen | 1.4 -> 1.8 migration, dialplan and configuration review, and other feature development |
| 00:01.27 | citywok | that sounds like fun :p |
| 00:01.29 | *** join/#asterisk Smirker (~x@124-254-82-70-static.bb.ispone.net.au) |
| 00:02.19 | Smirker | hey hey. is there any way to set a SIP header in an Asterisk call file? i'm trying to set up auto-answer on the first leg of the call. |
| 00:02.42 | leifmadsen | SIP_HEADER() ? |
| 00:02.47 | citywok | Smirker: i use SIPAddHeader(Call-Info:<sip:>\;answer-after=0) |
| 00:02.52 | leifmadsen | or SipAddHeader() |
| 00:03.00 | leifmadsen | citywok: well played |
| 00:03.07 | Smirker | thanks. but in a call file? |
| 00:03.17 | citywok | oh, no idea, i don't use call file's. i originate via AMI |
| 00:03.24 | citywok | s/file's/files/ |
| 00:03.47 | citywok | leifmadsen: :D |
| 00:04.11 | Smirker | ah okay. i think that's what i'll have to do! thanks |
| 00:04.32 | citywok | leifmadsen: who do you work for now? |
| 00:04.40 | leifmadsen | citywok: CoreDial |
| 00:04.42 | citywok | Smirker: i'm sure it can be done via call files, i just have no use for them |
| 00:05.16 | Smirker | there seems to be no parameter for it :( |
| 00:06.03 | citywok | ah, well, then do it in the dialplan the callfile calls |
| 00:06.38 | Smirker | afaik, the 'Channel' paramater acts as a Dial() command in the call file. once it has been answered, the dialplan executes. |
| 00:06.57 | [TK]D-Fender | So choose a better Channel |
| 00:06.59 | Smirker | unless there's a way to specify headers as a paramater in a Dial command, but i haven't discovered such thing |
| 00:07.02 | citywok | i dial the internal leg of the call via local/extension@internal-auto-answer and the context takes care of the sipheader. |
| 00:07.11 | citywok | Smirker: i think you are looking for local channels |
| 00:07.28 | Smirker | ah :) this sounds promising |
| 00:08.01 | leifmadsen | Local channels are FTW |
| 00:11.08 | citywok | what's a local channel? |
| 00:11.29 | leifmadsen | a very close channel |
| 00:11.47 | citywok | what's a private channel? |
| 00:11.47 | leifmadsen | there are also international and long-distance channels, but those cost money |
| 00:11.59 | leifmadsen | citywok: private channel is like a private cloud |
| 00:12.19 | citywok | what's a private cloud? |
| 00:12.29 | leifmadsen | you don't even want to know |
| 00:12.39 | citywok | what's an i don't even want to know? |
| 00:12.45 | citywok | okay, i'm done |
| 00:12.55 | WIMPy | leifmadsen: That reads as if you're writing about cause locations :-) |
| 00:13.05 | leifmadsen | :) |
| 00:14.09 | WIMPy | Which reminds me that such a variable might be quite usefull. |
| 00:18.22 | Smirker | citywok: thanks, local channels rock :) works perfectly |
| 00:18.41 | Smirker | i think i'll be abusing local channels hereafter |
| 00:21.20 | leifmadsen | Smirker: lots of people tend to once they figure out how to use them |
| 00:22.22 | citywok | Smirker: yw, and yes, as leifmadsen said, they do get heavily abused so to speak |
| 00:23.04 | leifmadsen | check out Asterisk: The Definitive Guide for a neat find-me-follow-me usage of Local channels to scatter timeouts for Dial() |
| 00:23.19 | citywok | haha self plugging eh!?!? :p |
| 00:23.24 | leifmadsen | maybe :) |
| 00:23.30 | leifmadsen | book is free, so hey :) |
| 00:23.36 | citywok | :p |
| 00:23.43 | leifmadsen | www.ateriskdocs.org |
| 00:23.52 | leifmadsen | EOP |
| 00:23.59 | citywok | s/ateriskdocs.org/asteriskdocs.org |
| 00:24.07 | leifmadsen | ya that |
| 00:24.30 | citywok | i forgot my trailing /, i wonder if you could correct that. lol |
| 00:24.45 | leifmadsen | hello |
| 00:24.49 | leifmadsen | s/hello/goodbye |
| 00:24.58 | citywok | s/s\/ateriskdocs.org\/asteriskdocs.org/test/ |
| 00:25.05 | citywok | apparently not :( |
| 00:25.09 | leifmadsen | s/goodbye/goodbye\// |
| 00:25.24 | leifmadsen | no escaping I guess |
| 00:25.28 | citywok | yea, sad panda |
| 00:25.34 | leifmadsen | that could get a bit crazy |
| 00:25.39 | leifmadsen | infinite loops ftw? :) |
| 00:25.43 | citywok | oh god |
| 00:26.21 | citywok | infobot'splosion |
| 00:27.50 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 00:46.03 | seanbright | so i have a server with some polycoms registered to it and mwi works fine |
| 00:46.12 | *** join/#asterisk FainaUkraina (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 00:46.18 | seanbright | i have a secondary server connected to the first with iax2 |
| 00:46.37 | seanbright | there is a phone registered to that server (also a polycom) that shares an 'extension' with the phones on the first server |
| 00:46.58 | seanbright | and i would like mwi to work there as well, except looking at the first server |
| 00:47.20 | citywok | eww distributed stuff... over my head! |
| 00:48.12 | leifmadsen | seanbright: I think you need distributed device state -- look at ais.conf |
| 00:48.22 | leifmadsen | it'll handle the mwi transfer amongst servers |
| 00:49.07 | leifmadsen | otherwise you'll need to subscribe to MWI from another server using SIP directly |
| 00:49.24 | seanbright | i thought ais was only good for high speed connections |
| 00:49.34 | seanbright | this link is not 'fast' |
| 00:49.36 | leifmadsen | seanbright: it is, so you need XMPP if they are further distributed |
| 00:49.42 | seanbright | gotcha |
| 00:49.45 | leifmadsen | or... via sip.conf: |
| 00:49.53 | leifmadsen | mwi => 1234:password@mysipprovider.com/1234 |
| 00:50.01 | leifmadsen | which can likely get real messy |
| 00:50.07 | seanbright | aye |
| 00:50.32 | leifmadsen | so ya, you need the xmpp distributed device state stuff it sounds like |
| 00:50.48 | leifmadsen | I'm not aware of a way to distribute it over the IAX2 connection |
| 00:51.32 | WIMPy | Does IAX have MWi support at all? |
| 00:53.13 | *** join/#asterisk [Outcast] (~anonymous@64.119.130.114) |
| 00:53.19 | seanbright | leifmadsen: what conf file i am looking for with the xmpp mojo? |
| 00:53.50 | leifmadsen | seanbright: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceStates_id265044.html#DeviceStates_id265061 |
| 00:54.03 | seanbright | thank you, sir. |
| 00:54.26 | leifmadsen | seanbright: np! |
| 00:54.35 | leifmadsen | and now I'm done working for the evening and will go watch TV with my lovely wife |
| 00:54.58 | carrar | PICS!! |
| 00:57.20 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 00:59.41 | *** join/#asterisk oej_ (~olle@2001:16d8:cc57:1000:70fa:1761:ddd4:6329) |
| 00:59.56 | carrar | http://www.youtube.com/watch?v=MJwb_wEaW2M |
| 00:59.58 | carrar | haha |
| 01:06.21 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 01:07.47 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:07.47 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:08.42 | SeRi | p3nguin: I started the porting process |
| 01:11.16 | p3nguin | It will probably go quickly. I know my last one did. |
| 01:38.56 | *** join/#asterisk worstadmin (~worst@68-119-70-42.dhcp.mtgm.al.charter.com) |
| 01:39.45 | worstadmin | Does anyone know a service I can SIP link my asterisk with and provides a pstn number for people to dial in on? |
| 01:40.08 | worstadmin | I'd like to create a bridge that is cell phone and soft client reachable (you know, assuming the phoen isnt using voip) |
| 01:40.25 | p3nguin | ~itsp |
| 01:40.25 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 01:40.42 | worstadmin | Thanks |
| 01:40.47 | worstadmin | Any particular one you guys prefer? |
| 01:41.10 | p3nguin | I use VoIP.ms primarily. |
| 01:41.36 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 01:42.25 | worstadmin | ~did |
| 01:42.25 | infobot | did is, like, Direct Inward Dialing, or just a phone number |
| 01:42.44 | kaushal | ~sms |
| 01:42.44 | infobot | [sms] Surprise mindsecks! |
| 01:44.03 | lanning | heh... is that like secksting? :) |
| 01:44.33 | coppice | Simple Minded Statements |
| 01:44.58 | kaushal | Hi |
| 01:45.08 | kaushal | I have questions about http://www.justdial.com/8888888888/phonesearch.php |
| 01:45.29 | kaushal | so it has both Voice as well as SMS application |
| 01:45.49 | kaushal | I have question about SMS application |
| 01:46.15 | kaushal | I suppose Asterisk cannot handle SMS Application ? |
| 01:46.46 | kaushal | I happen to see http://www.ozekisms.com/index.php?owpn=319 |
| 01:47.41 | kaushal | Can i handle incoming and Outgoing SMS in the setup ? |
| 01:47.51 | kaushal | via * PBX |
| 01:48.14 | kaushal | correct me if i am not getting it right |
| 02:16.20 | Sean-Der | Whats the proper way to do an if elseif else? |
| 02:17.12 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 02:21.02 | *** join/#asterisk saxa (~sasa@host242-95-static.223-217-b.business.telecomitalia.it) |
| 02:28.55 | *** join/#asterisk Neptu (~Neptu@213.88.195.203) |
| 02:35.37 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 02:47.30 | *** join/#asterisk coppice (~coppice@m121-202-49-128.smartone.com) |
| 02:55.18 | p3nguin | sean-der: You could do it in an AGI or a combination of a couple GotoIf()s. |
| 02:55.44 | Sean-Der | p3nguin, I ended up using GotoIf and had it jump to a context |
| 02:56.03 | Sean-Der | I still miss my brackets though |
| 02:59.46 | p3nguin | With GotoIf, you can go to a priority/label in the current extension in the current context, a different extension and priority in the current context, or a different context, extension, and priority. |
| 03:00.43 | *** join/#asterisk troyt (~troyt@c-24-10-222-127.hsd1.ut.comcast.net) |
| 03:43.33 | *** join/#asterisk gajini (~root@61.12.17.170) |
| 03:57.06 | *** join/#asterisk radic (~radic@dslb-178-002-229-035.pools.arcor-ip.net) |
| 04:02.17 | *** join/#asterisk ketas (~ketas@2001:ad0:91f:0:290:27ff:fea6:a66a) |
| 04:04.49 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 04:09.40 | *** join/#asterisk ketas (~ketas@2001:ad0:91f:0:290:27ff:fea6:a66a) |
| 04:12.49 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 04:18.34 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 04:19.06 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 04:25.03 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 04:39.15 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 05:00.42 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-150-222.dsl.stlsmo.sbcglobal.net) |
| 05:01.19 | LemensTS | If you set CALLERID(name)=something right before you dial out to your sip provider, and it does not show up on the receiving phone, there really is nothing left to do to fix that right |
| 05:07.27 | kaldemar | LemensTS: it's not usually the name part you want to set if you're talking about PSTN. |
| 05:12.49 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 05:38.54 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 05:44.21 | LemensTS | kaldemar: CALLERID(num) works fine. Im wanting to also set the name. |
| 05:50.15 | p3nguin | If it is actually relevant to set the name, which is normally discarded by the telco, you set it the same way you set the number only: Set(CALLERID(all)=Your Name <3145551212>) |
| 05:50.47 | p3nguin | You could also set the CALLERID(name) and CALLERID(num) separately if you like to complicate things. |
| 05:51.18 | kaldemar | LemensTS: where are you expecting to see the name? |
| 05:52.06 | p3nguin | Remembe that caller id name is looked up by the destination's phone company; caller id name is not something that is transmitted from the source phone like the number is. |
| 06:08.35 | [TK]D-Fender | p3nguin: it is actually... |
| 06:08.48 | [TK]D-Fender | p3nguin: It is also often disregarded, but that is another matter |
| 06:09.04 | [TK]D-Fender | p3gI've experienced telcos here where I had free reign |
| 06:18.51 | *** join/#asterisk k-man (~k-man@unaffiliated/k-man) |
| 06:20.11 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-qtwqyjaxsrqwmhcg) |
| 06:24.21 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-yhuysuxlnnhzvmgi) |
| 06:30.33 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 06:37.50 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 07:06.06 | *** join/#asterisk ven0m (~ven@87.69.145.169) |
| 07:06.59 | ven0m | Hi. I am trying to connect with Xlite to my Asterisk Server. the calls seems to be connec, but there is no audio and the call drops after few seconds. In the console, i get " Retransmission timeout reached on transmission" |
| 07:08.16 | kaldemar | ~sipnat |
| 07:08.17 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 07:18.48 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:21.52 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:27.06 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 07:30.50 | *** join/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 07:31.43 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 07:40.53 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:40.55 | schmidts | good morning |
| 07:44.31 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 07:49.34 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 07:55.04 | ChannelZ | blah! |
| 07:59.41 | *** join/#asterisk qakhan (~qakhan@182.185.179.231) |
| 08:00.08 | *** join/#asterisk stix (~stix@193.89.191.209) |
| 08:08.29 | *** join/#asterisk lftsy (~lftsy@194.38.160.113) |
| 08:15.03 | *** join/#asterisk waschtl (~waschtl@HSI-KBW-078-043-090-014.hsi4.kabel-badenwuerttemberg.de) |
| 08:15.54 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:17.18 | *** join/#asterisk oej (~olle@87.96.134.129) |
| 08:20.35 | qakhan | hi all |
| 08:21.34 | qakhan | i am using Digium AEX410P card. i setup an ivr on incoming call, press 1 for manager ext and press 0 for operator |
| 08:22.17 | qakhan | when caller press 1 card takes it 11, i noticed it in asterisk cli |
| 08:22.36 | qakhan | is there any DTMF setting for this card? |
| 08:27.52 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 08:30.49 | *** join/#asterisk amaninacan (~pie@unaffiliated/amaninacan) |
| 08:33.53 | *** join/#asterisk cmendes0101 (~nn@pool-173-58-50-52.lsanca.fios.verizon.net) |
| 08:36.24 | cmendes0101 | I'm having trouble getting callerid to work when sending a call. Can someone checkout the sip debug and see if anything is not correct. They are allowing that number to be used for callerid. http://pastebin.com/cAacLf40 |
| 08:37.08 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 08:41.25 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:48.40 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 08:50.15 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 08:50.29 | jacc0 | hi all!! good morning!! |
| 08:51.02 | jacc0 | does anyone know if asterisk 10 solves the issue of not being able to do macros 5 levels deep? |
| 08:51.10 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:528:6b19:5ff:e5a9) |
| 08:51.24 | jacc0 | (or 7 levels deep) |
| 08:53.14 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:56.21 | kaldemar | cmendes0101: your callerid seems to be set to 9095553333. |
| 08:58.20 | cmendes0101 | yah the provider said that would work to do a small test but I guess that could be the problem also if they didnt actually allow that. Have to wait till morning to hear back from support but just wanted to see if the rest of the sip appears ok |
| 09:03.05 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 09:19.06 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 09:20.01 | *** join/#asterisk prasanthprabalan (~root@218.248.24.19) |
| 09:21.44 | *** join/#asterisk irroot (~gregory@196-215-57-38.dynamic.isadsl.co.za) |
| 09:21.51 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
| 09:23.50 | ollii | printer management in linux mint 12 is a pain -.- |
| 09:37.10 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 09:37.23 | qakhan | i am using Digium AEX410P card. i setup an ivr on incoming call, press 1 for manager ext and press 0 for operator |
| 09:37.30 | qakhan | when caller press 1 card takes it 11, i noticed it in asterisk cli |
| 09:37.37 | qakhan | is there any DTMF setting for this card? |
| 09:39.51 | ollii | relaxdtmf in chan_dahdi.conf could be your setting |
| 09:40.05 | ollii | http://www.voip-info.org/wiki/view/Asterisk+DTMF |
| 09:40.10 | ollii | http://www.voip-info.org/wiki/view/chan_dahdi.conf |
| 09:46.08 | kaldemar | that chan_dahdi.conf page should be removed or at least replaced with links to samples for known versions. |
| 09:53.47 | kaldemar | qakhan: how did you verify the 11? |
| 09:54.23 | kaldemar | qakhan: you should enable dtmf debugging in logger.conf for the console line. you'll see what asterisk interprets. |
| 09:57.47 | *** join/#asterisk assalino (53f49842@gateway/web/freenode/ip.83.244.152.66) |
| 09:57.59 | assalino | hey guys |
| 09:58.05 | assalino | we setup a server running asterisk |
| 09:58.11 | assalino | and configured it to use google voice |
| 09:58.24 | assalino | now we have no idea of how to tell asterisk to call a number |
| 09:58.27 | assalino | is it a command line thing? |
| 09:58.35 | assalino | is there a "dial xxx" command? |
| 09:59.29 | kaldemar | depends. if you have chan_oss, chan_alsa or chan_console loaded you can dial with "console dial". otherwise, use a phone or CLI originate. |
| 10:00.36 | WIMPy | s/CLI/channel/ |
| 10:03.03 | kaldemar | the feature is caller CLI originate. yes, the command is "channel originate ...". |
| 10:12.34 | *** join/#asterisk assalino (53f49842@gateway/web/freenode/ip.83.244.152.66) |
| 10:13.06 | *** join/#asterisk Nasga (~Nasga@239.212.10.93.rev.sfr.net) |
| 10:18.28 | assalino | right, so if I type "console dial" |
| 10:18.35 | assalino | (considering I'm trying to call through google voice |
| 10:18.38 | assalino | this is what I get: |
| 10:18.39 | assalino | [Jan 11 10:18:06] WARNING[15202]: chan_oss.c:489 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory [Jan 11 10:18:06] NOTICE[15202]: console_video.c:133 console_video_start: voice only, console video support not present << Console call has been answered >> [Jan 11 10:18:08] WARNING[15203]: chan_oss.c:489 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory |
| 10:19.15 | assalino | it says answered, but then I get loads of warnings |
| 10:19.17 | assalino | with the same message |
| 10:19.25 | assalino | WARNING[15203]: chan_oss.c:489 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory |
| 10:20.39 | *** join/#asterisk oej (~olle@87.96.134.129) |
| 10:24.08 | *** join/#asterisk master_of_master (~master_of@p57B52E1B.dip.t-dialin.net) |
| 10:24.25 | kaldemar | assalino: looks like you don't have the choice to use "console dial". |
| 10:27.05 | *** join/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 10:28.18 | assalino | what should I use then? |
| 10:30.24 | *** join/#asterisk jkroon (~jkroon@dsl-241-253-235.telkomadsl.co.za) |
| 10:30.41 | FlashDeluxe | hi@all! i got a problem, i am using chan_capi 1.1.3 and asterisk 1.6.1 and i got two isdn cards with one controller each in the server. Now, if i want to call more than two users at the same time, the call failes :( can somebody help me please? i got my capi.conf and the error messages pasted here: http://paste.debian.net/151840/ |
| 10:35.39 | assalino | kaldemar, would you recommend any alternative? |
| 10:36.06 | *** join/#asterisk waschtl (~waschtl@HSI-KBW-078-043-090-014.hsi4.kabel-badenwuerttemberg.de) |
| 10:41.14 | kaldemar | assalino: cli originate or a phone, like i said. |
| 10:42.29 | assalino | sorry, i had to close and reopen the window |
| 10:42.32 | assalino | I get: No such command 'cli originate |
| 10:42.51 | assalino | nevermind, I removed the "cli" |
| 10:46.00 | assalino | kaldemar, it's weird though. I already have a dial plan setup. isn't there a way of just executing that dialplan? |
| 10:50.42 | *** join/#asterisk oej (~olle@87.96.134.129) |
| 10:51.14 | *** join/#asterisk tamiel (~tamiel@85-171-170-252.rev.numericable.fr) |
| 10:54.06 | jacc0 | console dial 123@yourcontext |
| 10:54.15 | jacc0 | that sould do it for you assalino |
| 10:54.26 | jacc0 | or take a look at: |
| 10:54.46 | assalino | hm, let me try |
| 10:54.46 | jacc0 | channel originate (not cli originate) |
| 10:55.52 | jacc0 | type this at CLI: 'core show help channel originate' |
| 10:56.23 | assalino | I still get the same chan_oss.c error |
| 10:56.24 | assalino | bummer |
| 10:56.42 | qakhan | kaldemar i setup in IVR if anyone press any digit expect 1 and 0 invalid playback run |
| 10:58.19 | qakhan | i see in asterisk cli incoming call when call press 1 it shows invalid number "11" |
| 11:02.12 | kaldemar | assalino: sure, "channel originate Local/exten@context ..." |
| 11:03.18 | kaldemar | qakhan: give something real to look at. |
| 11:09.15 | qakhan | kaldemar please look at this |
| 11:09.18 | qakhan | Executing [s@outgoing:1] Answer("DAHDI/2-1", "") in new stack |
| 11:09.18 | qakhan | <PROTECTED> |
| 11:09.19 | qakhan | <PROTECTED> |
| 11:09.19 | qakhan | <PROTECTED> |
| 11:09.19 | qakhan | <PROTECTED> |
| 11:09.19 | qakhan | <PROTECTED> |
| 11:09.19 | qakhan | <PROTECTED> |
| 11:09.19 | qakhan | <PROTECTED> |
| 11:09.20 | qakhan | <PROTECTED> |
| 11:09.20 | qakhan | <PROTECTED> |
| 11:09.21 | qakhan | <PROTECTED> |
| 11:09.21 | qakhan | <PROTECTED> |
| 11:09.22 | qakhan | <PROTECTED> |
| 11:09.22 | qakhan | <PROTECTED> |
| 11:09.46 | kaldemar | ~pb |
| 11:09.46 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 11:09.52 | kaldemar | qakhan: use pastebin next time. |
| 11:10.04 | qakhan | ok |
| 11:10.15 | kaldemar | qakhan: enable dtmf in logger.conf. |
| 11:10.22 | qakhan | ok |
| 11:11.17 | qakhan | kaldemar plz tell me how |
| 11:12.19 | kaldemar | what is unclear? |
| 11:12.42 | qakhan | there is no dtmf enable in logger.conf |
| 11:14.05 | kaldemar | edit logger.conf, add ",dtmf" at the end of the line that begins with "console =>" and give command "logger reload" in CLI. |
| 11:15.12 | qakhan | i did console => dtmf |
| 11:15.15 | qakhan | right? |
| 11:15.24 | kaldemar | no... |
| 11:15.52 | qakhan | then? |
| 11:16.12 | kaldemar | do what i told you. or did you not have a console line in there that had other values? |
| 11:16.37 | qakhan | kaldemar sorry i did not get you |
| 11:17.26 | qakhan | i added console => dtmf in loger.conf and reload the logger |
| 11:17.35 | qakhan | is it correct? |
| 11:19.35 | kaldemar | if you put it in the right place in the right file and gave the right command, yes. but i have no idea what you did. maybe you editer loger.conf instead of logger.conf or put it in the wrong place or gave a wrong command. who knows. |
| 11:21.07 | qakhan | kaldemar i got this |
| 11:21.40 | qakhan | <PROTECTED> |
| 11:21.41 | qakhan | <PROTECTED> |
| 11:21.41 | qakhan | <PROTECTED> |
| 11:21.41 | qakhan | [Jan 11 00:14:47] DTMF[2080]: channel.c:2385 __ast_read: DTMF end '1' received on DAHDI/1-1, duration 0 ms |
| 11:21.41 | qakhan | [Jan 11 00:14:47] DTMF[2080]: channel.c:2440 __ast_read: DTMF end accepted without begin '1' on DAHDI/1-1 |
| 11:21.41 | qakhan | [Jan 11 00:14:47] DTMF[2080]: channel.c:2451 __ast_read: DTMF end passthrough '1' on DAHDI/1-1 |
| 11:21.42 | qakhan | [Jan 11 00:14:47] DTMF[2080]: channel.c:2385 __ast_read: DTMF end '1' received on DAHDI/1-1, duration 0 ms |
| 11:21.42 | qakhan | [Jan 11 00:14:47] DTMF[2080]: channel.c:2440 __ast_read: DTMF end accepted without begin '1' on DAHDI/1-1 |
| 11:21.42 | qakhan | [Jan 11 00:14:47] DTMF[2080]: channel.c:2451 __ast_read: DTMF end passthrough '1' on DAHDI/1-1 |
| 11:21.42 | qakhan | <PROTECTED> |
| 11:21.43 | qakhan | <PROTECTED> |
| 11:21.47 | jacc0 | ~pb |
| 11:21.47 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 11:22.18 | jacc0 | qakhan: use pastebin for everything more then 3 lines |
| 11:22.36 | qakhan | jacc0 what is pastebin? |
| 11:22.51 | jacc0 | http://www.pastebin.com |
| 11:22.58 | kaldemar | jesus christ. it says what it is right there. |
| 11:22.59 | jacc0 | you can past your cli output there |
| 11:24.05 | qakhan | yes kaldemar what happend? |
| 11:24.40 | qakhan | but caller press 1 only once |
| 11:26.58 | qakhan | kaldemar u there? |
| 11:27.03 | jacc0 | 'duration 0 ms' sounds strange |
| 11:27.54 | qakhan | jacc0 is there card issue? |
| 11:27.57 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 11:28.35 | jacc0 | i can not answer that |
| 11:28.42 | jacc0 | i have no clue |
| 11:28.48 | qakhan | ok |
| 11:29.09 | jacc0 | maybe you should check region settings of your card |
| 11:29.15 | qakhan | ok |
| 11:29.30 | qakhan | kaldemar can u answer this |
| 11:31.44 | kaldemar | you should probably turn off hardware dtmf detection on the card. |
| 11:32.43 | qakhan | i just enable relaxdtmf=yes |
| 11:34.56 | assalino | kaldemar, would you happen to know an answer for this? http://bit.ly/zIHtp1 |
| 11:37.07 | *** join/#asterisk oej (~olle@87.96.134.129) |
| 11:37.42 | kaldemar | assalino: there is no simple answer for that. |
| 11:38.00 | assalino | I figured |
| 11:38.15 | assalino | all I really wanted was an API service that would do the job easily |
| 11:40.40 | kaldemar | you won't be able to do that with a few commands. for the making calls with asterisk, originate is the keyword. start with a softphone. using google voice is climbing a tree ass first. |
| 11:41.11 | assalino | I actually did that 1st |
| 11:41.23 | assalino | and I could get the call working |
| 11:41.29 | assalino | but I was doing softphone to softphone |
| 11:41.35 | assalino | using asterisk as a server |
| 11:41.51 | assalino | I figured since that wasn't the way I'd be using it, because I want the server to call the user |
| 11:42.01 | kaldemar | then learn to use originate with a soft phone. |
| 11:42.03 | assalino | there was no point in testing softphone to softphone |
| 11:42.17 | kaldemar | if you want to do it from a website, your best choice is probably AMI originate. |
| 11:42.27 | assalino | hm |
| 11:42.33 | kaldemar | ~ami |
| 11:42.33 | infobot | AMI is the Asterisk Manager Interface, a way to control an Asterisk server (and retrieve information) via a TCP/IP socket. More information is available at http://ofps.oreilly.com/titles/9780596517342/asterisk-AMI.html and http://voip-info.org/wiki/view/Asterisk+manager+API |
| 11:42.45 | assalino | nice, let me have a look at that :) |
| 11:43.23 | assalino | is the AMI an add-on? |
| 11:45.04 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 11:47.10 | kaldemar | assalino: no, it's a core feature. you just need to enable it in manager.conf. |
| 11:47.13 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 11:47.20 | assalino | right, on it |
| 11:47.45 | *** join/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 11:47.45 | *** join/#asterisk netman (netman@214.162.76.188.dynamic.jazztel.es) |
| 11:47.45 | *** join/#asterisk Tim_Toady (~fuzzy@193.92.236.29.dsl.dyn.forthnet.gr) |
| 11:47.45 | *** join/#asterisk gaetronik (~gaetan@ks370400.kimsufi.com) |
| 11:47.45 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 11:47.45 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 11:47.45 | *** join/#asterisk freeedrich| (friedrich@2a01:4f8:130:2023:1:151:0:babe) |
| 11:50.01 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 11:52.03 | assalino | kaldemar, could I just use the Asterisk GUI instead? |
| 11:53.07 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 11:53.14 | qakhan | kaldemar u there |
| 12:01.27 | kaldemar | assalino: it won't do that for you. |
| 12:02.12 | qakhan | kaldemar i just enabled relaxdtmf as jacc0 asked me to do that. now i disabled it again |
| 12:02.23 | qakhan | and still i m getting the same issue |
| 12:03.34 | kaldemar | qakhan: ask digium how to disable hardware dtmf detection on the card if that is what causes the duplicates. |
| 12:04.21 | qakhan | i am using Digium AEX410P |
| 12:07.37 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 12:13.00 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:13.00 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:13.02 | *** join/#asterisk s[X] (~s_x_@59.167.157.96) |
| 12:13.07 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 12:15.53 | *** join/#asterisk FluxiFlax2022 (~ali.jawad@193.227.186.146) |
| 12:19.33 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 12:20.49 | FluxiFlax2022 | Hi On 1.4 I had a scrip written to stop calls after 3hrs, I did change the script to use it on 1.8 using hangup. The problem is that on 1.8 I have in core show channels calls in RING and DOWN state for hundreds of hours, I need a command to drop those, is this a bug ? Or did I miss something ..example : |
| 12:21.24 | *** join/#asterisk irroot (~gregory@196-215-57-38.dynamic.isadsl.co.za) |
| 12:21.28 | FluxiFlax2022 | SIP/1xxxxxxx-0 default 20121899638 1 Down AppDial (Outgoing Line) 20121899638 762:14:0 1399775991 (None) |
| 12:21.29 | FluxiFlax2022 | SIP/xxxxxxxxx default |
| 12:22.03 | *** join/#asterisk skrusty (~ksrusty@62.252.24.138) |
| 12:22.13 | skrusty | afternoon |
| 12:23.06 | FluxiFlax2022 | SIPxxxxxxxxxxxx default 008801817672877 1 Ringing AppDial (Outgoing Line) 008801817672877 102:24:3 |
| 12:23.14 | skrusty | i am trying to connect a channel to ChanSpy using AMI Originate, but i get an error; i suspect this is because ChanSpy requires an answered channel (the source of the originate). Am i right in thinking this? |
| 12:23.15 | FluxiFlax2022 | 1.8.7.1 to be exact |
| 12:24.03 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 12:24.34 | *** join/#asterisk Faithful (~Faithful@202.189.73.144) |
| 12:25.17 | FlashDeluxe | hi! i got a problem with chan capi and asterisk, i have two isdn cards but only one b channel is used for each card :( does anybody got a suggestion on it? heres my config http://paste.debian.net/151840/ |
| 12:30.58 | WIMPy | I see both progress and proceeding before the busy. That looks like the call went out and was busy elsewhere. |
| 12:31.01 | *** join/#asterisk blitzrage (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:31.01 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 12:32.20 | *** join/#asterisk binbash_ (~peter@a80-127-250-64.adsl.xs4all.nl) |
| 12:32.36 | WIMPy | BTW: Are you sure you want the people you call to transfer you somewhere else on your system? |
| 12:36.35 | FlashDeluxe | i know, that the call-partner isn`t busy |
| 12:37.20 | FlashDeluxe | and i took a default config, i first want to get everythings working and after that i will configure it better ;) |
| 12:37.23 | WIMPy | Enable more debug. |
| 12:38.15 | FlashDeluxe | ok, one second please |
| 12:38.16 | kaldemar | FluxiFlax2022: TIMEOUT(absolute)=10800 will limit all your calls to 3 hours. |
| 12:38.32 | kaldemar | FluxiFlax2022: in an appropriate place in the dialplan ofcourse. |
| 12:39.21 | FluxiFlax2022 | kaldemar, will this take care of calls in DOWN and RING state ? |
| 12:40.23 | kaldemar | FluxiFlax2022: everything. |
| 12:46.28 | skrusty | does anyone know what versions of asterisk currently implement Dynamic parking lots? |
| 12:46.56 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:47.03 | skrusty | i see a patch has been applied as part of issue 0015135, but not sure how to determine from it which versions of asterisk contain this patch |
| 12:48.45 | *** join/#asterisk Ferraz (~Ferraz@189.84.174.173) |
| 12:49.31 | FlashDeluxe | WIMPy: Here we go http://paste.debian.net/151849/ |
| 12:50.28 | FluxiFlax2022 | kaldemar, will it send a BYE to the the caller and properly terminate the call or just cut it on Asterisk ? |
| 12:51.40 | FlashDeluxe | WIMPy and my extensions.conf and sip.conf http://paste.debian.net/151850/ |
| 12:54.53 | WIMPy | Wow. that's pretty unreadable stuff. |
| 12:55.22 | kaldemar | FluxiFlax2022: it will hang up properly. |
| 12:55.44 | WIMPy | But I see a cahnnel identification (which isn't further decoded unfortunatly) which also suggests that the call has been set up. |
| 12:56.03 | WIMPy | Otherwise that log doesn't contain any additional information. |
| 12:56.38 | FlashDeluxe | ok, maybe the call has been set up but why do i get a "busy" |
| 12:56.44 | WIMPy | No further info on the cause, either. |
| 12:56.59 | WIMPy | I don;t know. Your log doesn't tell. |
| 12:57.51 | WIMPy | Do you have to use CAPI? Support seems a little thin there lately. |
| 12:58.21 | *** part/#asterisk Ferraz (~Ferraz@189.84.174.173) |
| 12:58.33 | FlashDeluxe | yes i have to.. :( capi support is very rare, i know |
| 12:59.09 | WIMPy | Look fotr other debugging options. |
| 12:59.42 | WIMPy | Might be easier (cheaper) to get some cheap hardware instead. |
| 13:01.08 | FlashDeluxe | no, its a aserver of one of my customers, i would have to drive to him in order to get the new card into the server |
| 13:01.20 | FlashDeluxe | but maybe its a bug in chan_capi |
| 13:01.20 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:01.36 | FlashDeluxe | i could try the current trunk version |
| 13:02.19 | WIMPy | It still looks like an external issue to me, but as stated, the Log isn't that informative. |
| 13:02.36 | *** join/#asterisk ikevin_ (~kevin@poirot1.pck.nerim.net) |
| 13:02.41 | ikevin_ | hello |
| 13:03.00 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 13:03.52 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 13:03.53 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 13:04.09 | FlashDeluxe | mhh i will try to find a better debug option |
| 13:04.39 | ikevin_ | i've a problem while i try to setup a B410 BRI card (using Dahdi), while i make an outbound or inbound call, people can heard me so i can't heard people (/etc/dahdi/system.conf: http://pastebin.ca/2101938 , /etc/asterisk/chan_dahdi.conf: http://pastebin.ca/2101939 , /etc/asterisk/dahdi-channels.conf: http://pastebin.ca/2101940 ) |
| 13:04.48 | ikevin_ | anyone have an idea please? |
| 13:05.01 | *** join/#asterisk davlefou (~david@unaffiliated/davlefou) |
| 13:05.06 | davlefou | Hi, |
| 13:06.10 | davlefou | is it better to have softphone in iax or sip with en trunk iax ? |
| 13:07.56 | WIMPy | FlashDeluxe: Just out of interest: What kind of card is it? |
| 13:08.35 | WIMPy | ikevin: What are you using on the other end? Are you sure that works? |
| 13:09.04 | WIMPy | One-way audio is extremely unlikely on tdm hardware. |
| 13:12.25 | *** join/#asterisk t_dot_zilla (~t_dot_zil@unaffiliated/t-dot-zilla/x-2830497) |
| 13:15.43 | ikevin_ | WIMPy: it's connected to 2 T0 france telecom, i've an IPBX panasonic who work fine with it |
| 13:15.53 | FlashDeluxe | WIMPy a primux card from gerdes ag |
| 13:16.12 | *** join/#asterisk Diffen (~diffen@80.78.212.242) |
| 13:16.37 | ikevin_ | why it's unlikely? |
| 13:17.14 | WIMPy | ikevin_: It should be impossible to get on-way audio by misconfiguration. |
| 13:17.57 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
| 13:18.26 | ikevin_ | if think i've a configuration problem, so i can't find where |
| 13:18.38 | WIMPy | FlashDeluxe: Not the 4s, by chance? |
| 13:19.28 | WIMPy | ikevin_: What are you using on the other end? Have you tried to configure an echo extension and call in to that? |
| 13:20.54 | ikevin_ | The card is connected to france telecom network using 2 T0/S0 links, i've made a test with an echotest, that same, i can heard the introduction so echotest won't work |
| 13:21.23 | ikevin_ | for the moment i just make test to make an inbound call to an IVR |
| 13:21.44 | ikevin_ | (using a SIP trunk work fine with the same IVVR) |
| 13:21.57 | WIMPy | FlashDeluxe: From what I see the Gerdes cards are all HFC-S based. So you shouldn't need CAPI. |
| 13:22.40 | WIMPy | ikevin_: Did you Answer() the call? |
| 13:23.06 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:24.07 | ikevin_ | nop, just do: playback => echo => Playback => wait => hangup . i try to add a Answer |
| 13:24.42 | ikevin_ | so, if i can heard the playback, that mean the communication is answered? |
| 13:25.11 | WIMPy | Not neccessarily, but unless you use the noanswer option to Playback, it should auto-answer. |
| 13:25.29 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:27.00 | ikevin_ | does dahdi_monitor can help me to know if my problem come from dahdi configuration or asterisk configuration? |
| 13:28.02 | FluxiFlax2022 | <PROTECTED> |
| 13:28.41 | WIMPy | ikevin_: It will visualize the audio going in and out. |
| 13:29.18 | ikevin_ | ok, and if i've a configuration problem on asterisk i'll see nothing more than asterisk log? |
| 13:29.23 | WIMPy | FluxiFlax2022: Do you use a timeout in your Dials? |
| 13:30.15 | kaldemar | FluxiFlax2022: not really based on what you've said so far. |
| 13:30.27 | FluxiFlax2022 | WIMPy, as in TIMEOUT(Dial) ? |
| 13:30.50 | WIMPy | ikevin_: I can't imagine how to configure such behaviour. |
| 13:31.03 | kaldemar | FluxiFlax2022: timeout as in this timeout: Dial(Technology/Resource[&Technology2/Resource2[&...]][,timeout[,options[,URL]]]) |
| 13:31.21 | WIMPy | FluxiFlax2022: No, Dial(..., timeout) |
| 13:32.11 | FlashDeluxe | WIMPy i need capi, the gerdes cards do have their own proprietary driver + chan_capi |
| 13:32.31 | FlashDeluxe | WIMPy i tested it once with misdn, but i wasn`t very stable |
| 13:32.51 | ikevin_ | WIMPy: ok |
| 13:33.32 | WIMPy | FlashDeluxe: That doesn't seem to make sense. It's the standard chip. It should either work or not. But probaly just work. |
| 13:35.28 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 13:35.43 | FlashDeluxe | WIMPy i have been a trainee at gerdes ag for three years, i tested it there and it works in general, but as i said it isn`t very stable and its not recommended |
| 13:36.03 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 13:37.13 | WIMPy | FlashDeluxe: misdn1 or misdn2? |
| 13:37.49 | FlashDeluxe | misdn2 |
| 13:37.55 | Diffen | Hello. If extension 100 calls 101 from an x-lite and extension 101 have Follow Me active to a cell phone. The number that shows in the cellphone are the DID for extension 101. Is it possible to do some changes so it will show the DID of extension 100? |
| 13:38.47 | WIMPy | FlashDeluxe: I find it hard to believe that it works partially. |
| 13:39.19 | WIMPy | I can see of the the b410p twhich has additional hardware, but the Gerdes things look bog standard. |
| 13:39.28 | WIMPy | Diffen: How do you call your mobile? |
| 13:41.51 | FlashDeluxe | WIMPy mybe something changed |
| 13:42.18 | FlashDeluxe | WIMPy its been a while that i tested it and i was not so happy with my tests |
| 13:43.57 | FlashDeluxe | WIMPy ahh it seems to be fully supported http://www.misdn.org/index.php/MISDN_v2_Hardware |
| 13:44.12 | WIMPy | I didn;t like the idea of having an extra application and that's certainly a big argument for embedded systems, but it has been the best solution for me so far. |
| 13:44.21 | Diffen | Wimpy the asterisk is calling my mobile using find me / follow me. Im not sure how you mean. |
| 13:44.59 | WIMPy | Diffen: What way do you call out? POTS? ISDN? ITSP? |
| 13:45.36 | Diffen | Wimpy im using sip. |
| 13:46.40 | WIMPy | Diffen: then you have to ask your ITSP if they will allow you to send foreign caller IDs. |
| 13:47.21 | *** join/#asterisk moos3 (~rgenthner@cpe-72-224-121-41.maine.res.rr.com) |
| 13:47.37 | Diffen | Wimpy hmm I guess the problem is in asterisk. the invite that goes out from my asterisk to pstn are sent from extension 101 to the mobile phone. not from extension 100. |
| 13:48.09 | Diffen | so the sip provider do right when it shows extension 101 did in my mobile. |
| 13:48.42 | WIMPy | Asterisk doesn't change Caller ID unless you tell it to. |
| 13:48.55 | WIMPy | havong said that, I never used Followme. |
| 13:49.14 | WIMPy | grabs a hammer to flatten that o. |
| 13:50.35 | Diffen | wimpy ok. as far as i have understand how asterisk works is when extension 100 calls extension 101 to is 101 and from is 100 and thats cool. but when using follow me its not the extension 100 that calls my mobile, its extension 101. therefore my problem. |
| 13:53.33 | *** join/#asterisk DaneoShiga (~dshiga@kraz.dreamhost.com) |
| 13:55.21 | DaneoShiga | could someone tell me if using CDR(userfield)=#blah and recordagentcalls=yes on agents.conf will somewhat append the data like :"agent-3403-1318583256-2748625.gsm#blah" ? |
| 13:59.03 | [TK]D-Fender | DaneoShiga, No. Uou only get what you set |
| 13:59.07 | [TK]D-Fender | You* |
| 14:02.08 | DaneoShiga | got it... gonna imagine some other way to do that, thanks [TK]D-Fender |
| 14:05.41 | *** join/#asterisk serafie (~erin@nat/digium/x-tbxtfukiylecwesh) |
| 14:07.30 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 14:08.20 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 14:09.36 | DaneoShiga | recordagentcalls already insert some data in userfield... i need to figure out a easy way to append more data there. |
| 14:12.36 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:20.29 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 14:24.38 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ylhprtzzbawpiatk) |
| 14:24.38 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:28.17 | moos3 | so have the really crazy idea, installing asterisk 1.8 on http://www.viaembedded.com/en/products/boards/productDetail.jsp?productLine=1&id=1670&tabs=1 should work right ? |
| 14:29.15 | pabelanger | moos3: which distro of linux? |
| 14:29.38 | moos3 | pabelanger not sure yet, i'm leaning toward archlinux to just keep it extremely slim |
| 14:29.57 | WIMPy | I don't think any of the console channels support video so the Full-HD video output may stay unused :-) |
| 14:30.27 | pabelanger | moos3: You'll have to try, but I think you'll be fine |
| 14:31.06 | moos3 | yea i want a simple way to give my pops a sip interface for a WIP with out taking up a bunch of space, i think this will work |
| 14:34.40 | [TK]D-Fender | WIP? |
| 14:34.49 | *** join/#asterisk waschtl (~waschtl@HSI-KBW-078-043-090-014.hsi4.kabel-badenwuerttemberg.de) |
| 14:36.35 | akrohn | maybe a WIP310, which is a cisco wireless sip phone |
| 14:41.38 | *** join/#asterisk mattsqz (~luser@67-61-162-124.cpe.cableone.net) |
| 14:44.03 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 14:46.29 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 15:04.35 | *** join/#asterisk darKxyde (c1308404@gateway/web/freenode/ip.193.48.132.4) |
| 15:04.41 | darKxyde | Hello there |
| 15:06.17 | darKxyde | I'm having issues with a Patton gateway, does anyone can help? |
| 15:06.20 | *** join/#asterisk Neptu (~Neptu@213.88.195.203) |
| 15:07.55 | *** join/#asterisk irroot (~gregory@196-215-57-38.dynamic.isadsl.co.za) |
| 15:08.25 | *** part/#asterisk irroot (~gregory@196-215-57-38.dynamic.isadsl.co.za) |
| 15:08.44 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 15:09.34 | [TK]D-Fender | darKxyde, Show us what is happening and we'll see what we can advise on it. |
| 15:09.37 | *** join/#asterisk mattsqz (~luser@67-61-162-124.cpe.cableone.net) |
| 15:09.38 | [TK]D-Fender | ~pb |
| 15:09.38 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 15:09.40 | [TK]D-Fender | ^^^ |
| 15:10.16 | darKxyde | Well, I'm trying to set it as a ISDN/SIP Gateway |
| 15:10.28 | darKxyde | my externals calls are perfectly working |
| 15:10.40 | darKxyde | but I can't find a way to make the incomings call working |
| 15:11.09 | darKxyde | Patton gets them |
| 15:12.16 | *** join/#asterisk BMJ (~bjohns@nat/digium/x-utoejbharnzrwoce) |
| 15:12.16 | *** mode/#asterisk [+o BMJ] by ChanServ |
| 15:12.22 | darKxyde | but it fails to "send it" to the sip network |
| 15:21.25 | [TK]D-Fender | darKxyde, Ok, this really is a Patton config issue. Wait around a bit. Ask again in a few hours if no-one has been able to help. Also I'd check with Patton's community resources and the mailing list. |
| 15:23.41 | darKxyde | I've looked for a Patton's community for days, all I've found is the support team on www.patton.com(or so) but they don't seem to be that reactive (2/3weeks) |
| 15:25.59 | [TK]D-Fender | darKxyde, That's pretty bad... |
| 15:30.42 | *** join/#asterisk ltd (~z@zx.io) |
| 15:32.16 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 15:34.34 | akrohn | i have an auto attendant that forwards callers to an employee cell phones. occasionally, the call drops after a minute or so. is there a setting i can change to minimize this behavior? |
| 15:38.13 | [TK]D-Fender | akrohn, There isn't some option like "dontdropmycalls=yes" |
| 15:38.24 | [TK]D-Fender | akrohn, You need to really look at the problem. |
| 15:38.54 | min3r | akrohn, im not sure in astrisk right off. I do know that freepbx has a 120 sec limit default before dropping calls |
| 15:39.35 | akrohn | [TK]D-Fender, that is the correct answer. but this stuff is so difficult to troubleshoot |
| 15:39.55 | [TK]D-Fender | FreePBX doesn't drop calls. Or place calls. Asterisk does. |
| 15:40.09 | min3r | yeah freepbx is a front end to asterisk config |
| 15:41.16 | [TK]D-Fender | akrohn, Sometimes, but so far we have nothing to base that on. |
| 15:42.00 | leifmadsen | dropped calls via SIP are typically due to NAT issues or network communications issues where a message isn't being responded to. |
| 15:42.38 | carrar | Just blame the network guy and then go to lunch |
| 15:42.51 | min3r | lol |
| 15:44.37 | [TK]D-Fender | I do my best blaming after lunch... |
| 15:44.48 | akrohn | thanks, i will check it out |
| 15:46.53 | leifmadsen | you're going to have to look at the console and review the SIP debug and such and see what it is doing and why it isn't working |
| 15:50.04 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 15:51.10 | *** join/#asterisk eZz (~ez@195.114.6.134) |
| 15:51.56 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:52.06 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 15:53.10 | SeRi | whats and LIDB/CNAM? |
| 15:59.31 | pabelanger | http://en.wikipedia.org/wiki/LIDB and http://en.wikipedia.org/wiki/CNAM |
| 16:02.58 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 16:02.58 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:03.08 | *** join/#asterisk tompaw (~tompaw@slave30.tesserakt.eu) |
| 16:04.22 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 16:04.31 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 16:19.31 | *** join/#asterisk tompaw (~tompaw@slave30.tesserakt.eu) |
| 16:20.05 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v004-093.mobile.uci.edu) |
| 16:21.09 | *** join/#asterisk hfb (~hfb@pool-98-119-109-145.lsanca.dsl-w.verizon.net) |
| 16:25.02 | rgsteele | [TK]D-Fender: Just FYI, I did get it to work this morning. Apparently udev picked up on the second module in the first card before the first module. |
| 16:25.10 | rgsteele | [TK]D-Fender: Thanks again for the suggestion. |
| 16:25.32 | [TK]D-Fender | rgsteele, Modul order in the card isn't a matter for udev... |
| 16:25.57 | [TK]D-Fender | IIRC the ports aren't orderd as one might think they should be on those older revision cards. |
| 16:26.54 | *** join/#asterisk dj_hamsta (~werwer@unaffiliated/dj-hamsta/x-2342346) |
| 16:27.02 | *** join/#asterisk lhfnet (~lhfnet@net-cdd-fw01.cddlasmercedes.com) |
| 16:27.05 | dj_hamsta | so some ops here are employed by digium ? |
| 16:27.46 | lhfnet | Hello All, I upgraded from DAHDI 2.5.1 to 2.6.0 but low the outside calls have a lot of noise, I downgrade the version to 2.5.1 again, but the noise remains. Any idea? |
| 16:28.04 | [TK]D-Fender | dj_hamsta, Most are. |
| 16:28.25 | dj_hamsta | that is awesome |
| 16:28.59 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:30.00 | lhfnet | no one? |
| 16:30.36 | rgsteele | [TK]D-Fender: Er, well, the port numbers are etched into the metal on the back of the card. |
| 16:31.11 | [TK]D-Fender | rgsteele, Well I've never heard of the ports being reported as individual devices like that. |
| 16:31.34 | [TK]D-Fender | rgsteele, order between 2 cards sure, but not ont he ports within a card |
| 16:34.17 | WIMPy | lhfnet: Sounds like https://issues.asterisk.org/jira/browse/DAHLIN-275 |
| 16:35.24 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
| 16:36.06 | lhfnet | WIMPy: Sorry I didn't restart asterisk and dahdi, with the 2.5.1 version the noise disappear |
| 16:36.29 | lhfnet | WIMPy: the problem is with 2.6.0 |
| 16:41.29 | rgsteele | Weird, Asterisk crashed, had to restart it, now I'm getting the weird silence thing again. |
| 16:41.40 | rgsteele | logs look normal though, same as when they work |
| 16:41.52 | rgsteele | It's like the port changed or something |
| 16:48.26 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 16:52.34 | *** join/#asterisk oej_ (~olle@2001:16d8:cc57:1000:c043:9096:39c1:cee7) |
| 16:52.42 | *** part/#asterisk lhfnet (~lhfnet@net-cdd-fw01.cddlasmercedes.com) |
| 16:54.58 | *** part/#asterisk amaninacan (~pie@unaffiliated/amaninacan) |
| 17:03.17 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 17:03.34 | *** join/#asterisk oej (~olle@87.96.134.129) |
| 17:04.34 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 17:12.34 | p3nguin | rgsteele: What "port" are you talking about? |
| 17:20.53 | *** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de) |
| 17:22.47 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 17:22.52 | *** join/#asterisk l2trace99 (~jr@rrcs-71-43-104-238.se.biz.rr.com) |
| 17:24.31 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
| 17:26.00 | WintermeW | hi. is it possible to allow only one single destination call for each peer ? i don't know how to implement that, i looked at other things like freeswitch, but i'm kindof lost.. :s |
| 17:29.27 | dj_hamsta | this asterisk gui is seriously easy |
| 17:30.14 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:38.38 | SeRi | dj_hamsta: I want to see you say the same thing when you make a hand edit on the files and go back to the gui... ;) |
| 17:42.03 | *** join/#asterisk irroot (~gregory@197.109.187.66) |
| 17:42.36 | *** part/#asterisk irroot (~gregory@197.109.187.66) |
| 17:47.37 | *** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de) |
| 17:48.21 | *** join/#asterisk Diffen (~diffen@90-231-44-70-no32.tbcn.telia.com) |
| 17:55.31 | WIMPy | WintermeW: That's what contexts are for. |
| 18:03.14 | *** join/#asterisk tompaw (~tompaw@slave30.tesserakt.eu) |
| 18:03.48 | *** join/#asterisk thansen (~thansen@c-67-177-32-87.hsd1.ut.comcast.net) |
| 18:04.50 | thansen | how can I see attempts to login to sip accounts server side? |
| 18:05.09 | [TK]D-Fender | thansen, enable SIP debug at * CLI |
| 18:06.04 | *** join/#asterisk [Outcast]_ (~anonymous@westford-nat.juniper.net) |
| 18:09.34 | thansen | [TK]D-Fender: awesome, thanks..I recently bounced a server and I have some clients that stopped connecting |
| 18:09.41 | thansen | what would cause... SIP/2.0 401 Unauthorized |
| 18:09.55 | p3nguin | Seems pretty clear to me. |
| 18:10.24 | thansen | unauthorized at what level...I have not changed a single password or anything |
| 18:10.53 | p3nguin | It means the device needs to authenticate because it is not authorized. |
| 18:11.18 | thansen | right, all of them have the same credentials as before |
| 18:11.46 | p3nguin | Then I guess it won't be much problem to get an authentication. |
| 18:12.17 | thansen | they're spread out geographically as well as what the clients types are (softphone, pap2t, etc) |
| 18:13.20 | [TK]D-Fender | thansen, And if you want to pin down your problem you will ignore all those other devices, all their configs, locations, etc and just focus on all the bits that are directly related to this one device |
| 18:14.21 | thansen | right, I've enabled debugging for the issue at my house and I have a pap2t and empthy/sofiasip...all 3 giving the same issue |
| 18:14.30 | thansen | or, the IP at my house |
| 18:14.38 | *** join/#asterisk sled-dog (~luser@65-124-95-55.dia.static.qwest.net) |
| 18:15.10 | [TK]D-Fender | thansen, If you are looking for our help you should pastebin your sip.conf along with the complete debug. |
| 18:15.15 | [TK]D-Fender | ~pb |
| 18:15.15 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 18:16.02 | p3nguin | Complete means you don't hide things like IP addresses and device names. Complete means the only thing hidden will be passwords in the conf. |
| 18:18.42 | *** join/#asterisk lodac (~lodac@71-83-12-142.static.aldl.mi.charter.com) |
| 18:19.58 | lodac | would I be able to use asterisk to route between exsiting nortel bcm's with t1 cards? |
| 18:20.42 | thansen | http://pastebin.com/iu5Ara90 there is a sample account that fails |
| 18:21.37 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:22.47 | Qwell | lodac: sure, why not? |
| 18:24.45 | [TK]D-Fender | thansen, That is less than half of what was requested |
| 18:24.46 | lodac | I have a situation, maybe you could take a look. https://docs.google.com/document/d/1ti4hpvLUgY9aq9YO51B17nFjnmCKEe_aZGZLe_mOm3Y/edit |
| 18:24.59 | thansen | http://pastebin.com/mwT1MutE |
| 18:25.12 | thansen | there's the sip debug sequence |
| 18:25.48 | thansen | the server changed IP addresses, would that have any effect on the authentication? |
| 18:25.57 | lodac | ICES, SEAS, SFHA are sites. the Opt11c is at the SFHA site. Opt11c is dying, along with ICES BCM. My idea is to replace the Opt11c with a 4port T1 asterisk box to continue using the SEAS and SFHA BCM's |
| 18:26.30 | lodac | replace ICES phones with SIP phones |
| 18:27.33 | lodac | which would use asterisk for voip |
| 18:30.49 | [TK]D-Fender | thansen, "sip show peer 1298" , "sip show settings" |
| 18:31.03 | cmendes0101 | Setting up a new server and getting this error from the settings on the provider in sip.conf [Jan 11 09:27:34] WARNING[17352]: chan_sip.c:3389 __sip_xmit: sip_xmit of 0x2aaaac0fd030 (len 536) to (providerip):5080 returned -1: Operation not permitted. Any idea what caused this? |
| 18:33.52 | wdoekes2 | cmendes0101: iptables rules in the OUTPUT chain? |
| 18:34.13 | thansen | [TK]D-Fender: http://pastebin.com/WEBzcgHm |
| 18:34.41 | thansen | wonders about.. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6a10603b" |
| 18:35.41 | [TK]D-Fender | thansen, So your server has a pubilc IP directly on it? |
| 18:35.46 | cmendes0101 | wdoekes2: ACCEPT tcp -- anywhere anywhere state NEW tcp dpt:sip & ACCEPT tcp -- anywhere anywhere state NEW tcp dpt:5080 |
| 18:36.09 | thansen | [TK]D-Fender: yep (a slew of them currently if that matters) |
| 18:36.18 | wdoekes2 | cmendes0101: udp? |
| 18:36.40 | cmendes0101 | ahhh thanks lol |
| 18:36.46 | Qwell | Why 5080? O.o |
| 18:36.50 | [TK]D-Fender | thansen, might pose a problem... * could be receiving on one IP and sending on another due to how packets are formed, and * is bound to all ports as well. |
| 18:37.03 | [TK]D-Fender | thansen, Multi-homed = servere PITA |
| 18:37.45 | *** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 18:37.45 | [TK]D-Fender | thansen, I would do a raw packet dump to very sure where things are really comeing from & going to... |
| 18:37.49 | cmendes0101 | Qwell: not sure 1 provider is showing that error the other provider is showing the same error but with 5060. Sip.conf is at 5060 also |
| 18:37.55 | thansen | I *think* it had multiple IPs previously, but I don't remember exactly |
| 18:38.07 | SeRi | p3nguin: can I call you? I need to test my CID. it's showing as uwknown for what ever reason... |
| 18:38.11 | thansen | maybe I can just start it up bound to one IP or does that not help? |
| 18:38.30 | [TK]D-Fender | thansen, won't help the outbound |
| 18:38.36 | thansen | nods |
| 18:38.49 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-edumixdjlzmscfos) |
| 18:38.54 | [TK]D-Fender | thansen, that is based on your OS's stack which per the norm could go wonky on you |
| 18:40.25 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:40.38 | thansen | lemme see if it had multiple before just to be more sure that's likely the issue... |
| 18:43.13 | rgsteele | p3nguin: I meant the module that it's sending audio to |
| 18:43.52 | thansen | [TK]D-Fender: ok, it had 3 previously and 5 now (minus internal) |
| 18:45.00 | [TK]D-Fender | thansen, Ok, Can't say if how it changed comes int play or not on this.... |
| 18:45.01 | thansen | is this standard... WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6a10603b" |
| 18:45.12 | [TK]D-Fender | thansen, Yup |
| 18:45.23 | *** join/#asterisk tompaw (~tompaw@slave30.tesserakt.eu) |
| 18:46.01 | *** join/#asterisk jkroon (~jkroon@dsl-241-253-235.telkomadsl.co.za) |
| 18:46.03 | thansen | ok, well, is there a nice little doc written up on how to gather the raw packet dump? I normally don't delve into that |
| 18:46.27 | [TK]D-Fender | thansen, Well this is conjecture as we haven't followed through on the packet trace. Lets back up a step |
| 18:46.34 | thansen | ok |
| 18:46.44 | [TK]D-Fender | thansen, Take a look at the realm options on what you're trying to have connect first |
| 18:46.48 | thansen | gives thanks for the help |
| 18:47.35 | cmendes0101 | wdoekes2: thanks again, just double checked and its fine now |
| 18:48.11 | [TK]D-Fender | thansen, I know Sofia is used by FS and I'm seeing places where their use of realm is the IP of a box(the phone itself?). Not sure of the full implications or relevance, but it came up as a visual difference on a trace on their WIKI for interconnection. |
| 18:48.41 | *** join/#asterisk FainaUkraina (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 18:49.50 | thansen | well, I don't see 'realm' in the interface for empathy accounts or for the pap2t |
| 18:51.34 | [TK]D-Fender | thansen, Ok, I'm afraid we're hitting the limits of my experience on this then.... |
| 18:53.47 | *** join/#asterisk DelphiWorld (~VoiceClou@openvpn/user/DelphiWorld) |
| 18:53.48 | thansen | I'm also getting this garbage too.. Received SIP subscribe for peer without mailbox: |
| 18:53.52 | *** part/#asterisk DelphiWorld (~VoiceClou@openvpn/user/DelphiWorld) |
| 18:56.16 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 19:02.16 | thansen | [TK]D-Fender: got it! |
| 19:02.34 | thansen | it is a multi-ip issue |
| 19:02.37 | [TK]D-Fender | :) |
| 19:02.57 | thansen | I have dns for sip. and it points to what will be a floating IP |
| 19:03.01 | [TK]D-Fender | thansen, Nice to know the outer reaches of my knowledge are occasionally useful :) |
| 19:03.21 | thansen | so it's the not the *mail* IP bound to the nic |
| 19:03.25 | thansen | *main* |
| 19:03.42 | thansen | I just hacked my client to point to the primary IP and it connected right away |
| 19:04.32 | *** join/#asterisk b3nt_pin (~b3ntpin@142.162.36.51) |
| 19:09.08 | thansen | [TK]D-Fender: this looks quite old, but probably still valid... http://lists.digium.com/pipermail/asterisk-dev/2004-July/005211.html |
| 19:10.30 | [TK]D-Fender | thansen, Well the writing is clear... guess its the rule until we have reason to believe otherwise... |
| 19:11.07 | thansen | yeah, it makes good sense now, I'm sure before my dns entry pointed to the primary IP on the nic |
| 19:13.59 | *** join/#asterisk ectospasm (~ectospasm@66.172.33.249) |
| 19:30.19 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-157-240.chyn.qwest.net) |
| 19:32.59 | *** join/#asterisk WintermeW (~toor@78.250.27.2) |
| 19:35.02 | *** join/#asterisk CcSsNET (~CcSsNET@c-98-216-138-179.hsd1.ma.comcast.net) |
| 19:35.36 | WintermeW | i don't really understand how the realtime dialplan is implemented..it says you must add 'switch => Realtime/mycontext@realtime_ext' , but then it's not fully realtime since you still have hardcoded statements in your extensions.conf file |
| 19:35.44 | CcSsNET | ok i am a voip noob. i have a unopened magic jack original sitting here. is there any hope for getting it to work fully on linux with or without asterisk? |
| 19:39.53 | *** join/#asterisk LemensTS (~matthew@99.160.252.235) |
| 19:40.47 | LemensTS | How would I pickup exten => s,n,Dial(SIP/800,21,tr) ??? exten = _**800,1,Pickup(${EXTEN:2}) does not work |
| 19:40.48 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 19:40.59 | LemensTS | I think its because it goes into 's' extension and not '800' |
| 19:42.42 | cmendes0101 | with mysql() is there any way to verify the connection is still active? noticing alot of server has gone away warnings but the connid is still set |
| 19:43.12 | [TK]D-Fender | LemensTS, "core show application pickup" |
| 19:44.01 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 19:48.30 | *** part/#asterisk LemensTS (~matthew@99.160.252.235) |
| 19:51.39 | *** join/#asterisk twanny796 (~twanny@46.11.127.88) |
| 19:52.16 | thansen | [TK]D-Fender: thanks again for the help! |
| 19:52.21 | *** part/#asterisk thansen (~thansen@c-67-177-32-87.hsd1.ut.comcast.net) |
| 19:53.33 | *** join/#asterisk n3hxs (~ed@ann100fw01.answernet.com) |
| 19:55.48 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 19:59.14 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
| 19:59.25 | *** part/#asterisk CcSsNET (~CcSsNET@c-98-216-138-179.hsd1.ma.comcast.net) |
| 20:00.57 | *** join/#asterisk tompaw (~tompaw@slave30.tesserakt.eu) |
| 20:05.04 | *** join/#asterisk wesphillips (~wphill04@c-76-30-173-12.hsd1.tx.comcast.net) |
| 20:07.10 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 20:10.26 | *** join/#asterisk wesphillips (~wphill04@192.160.117.129) |
| 20:11.29 | *** join/#asterisk MindTheGap (~caio@gw-infra.advance.tur.br) |
| 20:12.05 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 20:16.17 | MindTheGap | hello all... Suppose I have to manage a mid size call center. Is there any open source candidate that wont completely take over my asterisk and turn my dialplan into a mess? |
| 20:17.47 | pabelanger | o.0 |
| 20:19.06 | Qwell | MindTheGap: vim |
| 20:22.30 | [TK]D-Fender | MindTheGap, "candidate"? |
| 20:22.53 | [TK]D-Fender | What is a "candidate" that is "taking over Asterisk"? |
| 20:22.58 | ikevin | an alternative of asterisk for callcenter: http://www.freepbx.org/ |
| 20:23.36 | [TK]D-Fender | ikevin: how is FreePBX an ALTERNATE to Asterisk? |
| 20:23.41 | WintermeW | nobody to explain is it still necessary to have contexts in extensions.conf when using extensions realtime..it looks incoherent to me |
| 20:23.50 | WintermeW | why* |
| 20:24.04 | [TK]D-Fender | ikevin: that's like asking what is better ... a wheel .. or a car. |
| 20:24.42 | ikevin | freebpx is based on asterisk so optimised for a lot of call |
| 20:24.50 | [TK]D-Fender | ikevin: No |
| 20:25.05 | _Corey_ | ikevin: freepbx is one of the less appropriate things for call center applications, FWIW |
| 20:25.24 | [TK]D-Fender | ikevin: FreePBX is a configuration frontend for building * configs. It is not a substitute for it. |
| 20:26.15 | MindTheGap | ok i can get basic callcenter functionality straight from asterisk, just create extensions, queues, disciplines and ppl start taking calls... |
| 20:26.39 | ikevin | i speak about freepbx distribution, not the pbx daemon |
| 20:27.46 | MindTheGap | proble is managing it, offering supervisors an interface so he can push ppl back and forth among queues, follow sla, give agents a chance to pause (with motives!), and show all that stuff... |
| 20:28.10 | MindTheGap | generate reports, etcetera. |
| 20:28.24 | [TK]D-Fender | ikevin: A distribution is also not a "alternative" to an application. |
| 20:28.34 | [TK]D-Fender | ikevin: you keep comparing apples & oranges |
| 20:28.46 | MindTheGap | something like the switchvox SMB callcenter stuff. |
| 20:28.57 | _Corey_ | MindTheGap: I'm assuming that you've already investigated FreePBX based on the way you phrased your question ("turn dialplan into a mess")... unfortunately on the open-source level there isn't much else going on |
| 20:29.18 | MindTheGap | _Corey_, i see |
| 20:29.23 | _Corey_ | MindTheGap: What's your definition of "mid-sized" ? |
| 20:29.37 | MindTheGap | _Corey_, ~200 seat |
| 20:29.53 | _Corey_ | Too large for Switchvox then |
| 20:30.29 | [TK]D-Fender | Switchvox doesn't have a solution that can handle 200 seats? |
| 20:30.42 | _Corey_ | 200 seats yes, 200 concurrent calls no |
| 20:30.54 | ikevin | <[TK]D-Fender> ikevin: you keep comparing apples & oranges <=== orange is not an apple optimised |
| 20:31.00 | MindTheGap | _Corey_, do you know indosoft q-suite? does it worth it? |
| 20:31.28 | MindTheGap | SMB 355 tops at 75 concurrent calls. according to docs. |
| 20:31.40 | _Corey_ | MindTheGap: 200 seats puts you in real "contact center platform" territory... some play nicely with asterisk though |
| 20:31.50 | _Corey_ | I'm not familiar with Indosoft |
| 20:32.17 | [TK]D-Fender | ikevin: FreePBX ISO is a distro that includes *. You can't go and compare that to * as jsut a piece of software. it isn't an ALTERNATIVE. |
| 20:32.33 | [TK]D-Fender | ikevin: It contains *. Your comparison is inappropriate |
| 20:33.20 | _Corey_ | MindTheGap: For 200 seats with basic inbound and simple analytics needs, I'd probably recommend a straight Asterisk setup w/Queuemetrics (or similar) |
| 20:33.55 | ikevin | it contain * so optimized for a large amount of simultaneous calls |
| 20:34.16 | ikevin | so, i going to bed, have a good evening (else if someone has an idea: http://forums.digium.com/viewtopic.php?f=1&t=81251&sid=c82877b2b44503eb9f82f719f26f18dc) |
| 20:34.30 | _Corey_ | ikevin: FreePBX is *not* optimized for a large amount of simultaneous calls |
| 20:34.40 | Qwell | ikevin: FreePBX is the opposite of optimized for large call volume. |
| 20:34.49 | Qwell | disoptimized? |
| 20:35.19 | MindTheGap | _Corey_, queuemetrics is focused on analitics only isnt it? i still have to manage agents and give some autonomy to supervisors. |
| 20:35.31 | _Corey_ | MindTheGap: Yes, indeed |
| 20:35.33 | ikevin | i used it in my last job for ~100 150 simultaneous call it working perfect |
| 20:35.44 | Qwell | Oh, 100 whole calls. Shocking. |
| 20:36.16 | Qwell | Let us know when it can handle about 50x that. Then we can start talking about it being optimized. |
| 20:36.43 | _Corey_ | MindTheGap: for something more robust (read "enterprise grade") analytics, outbound (predictive), multi-channel etc... you an PM me for a recommendation if you want |
| 20:38.14 | ikevin | Qwell, i'm not working on large call center, just middle |
| 20:38.33 | _Corey_ | ikevin: A lot of factors determine just when FreePBX will cause a system to become unstable, but in the past it was as easy as putting 30-40 phones in a queue with a ringall strategy and putting just a couple calls in queue... |
| 20:39.25 | Netgeeks | anyone here got sip realtime working with ldap? |
| 20:39.33 | [TK]D-Fender | ikevin: FreePBX ISO contains the same boring * you install on your own. There is nothing changed there. There is no "optimization". |
| 20:41.03 | *** join/#asterisk voipeng (~voipeng@75-150-128-2-Philadelphia.hfc.comcastbusiness.net) |
| 20:42.00 | voipeng | hi, im tryin to install zaptel for the first time on a new server, got the kernel-devel package installed, and run ./configure fine, but i cannot make zaptel, http://pastebin.com/tM7L7cyS |
| 20:42.19 | Qwell | voipeng: Zaptel is dead. You need to use DAHDI. |
| 20:42.37 | voipeng | i cannot I use voiceaxis |
| 20:42.43 | [TK]D-Fender | Qwell, He's probably still working to migrate a decrepit 1.4 install |
| 20:42.46 | voipeng | yep |
| 20:42.57 | voipeng | not migrate just get it install at all on a new server |
| 20:42.58 | [TK]D-Fender | voipeng, And nobody knows that name BTW.... won't mean anything to them |
| 20:43.11 | voipeng | how should i refer to it? |
| 20:43.21 | _Corey_ | He's talking about Coredial's platform... ask leifmadsen |
| 20:43.24 | _Corey_ | :) |
| 20:43.37 | [TK]D-Fender | For globa reference : VoiceAxis = GUI distro w/ custom front-end, their own repos, etc, and stuck in the stone-age |
| 20:43.39 | Qwell | voipeng: Zaptel will never build on new kernels. |
| 20:44.39 | leifmadsen | did zaptel have menuselect? |
| 20:45.07 | voipeng | i was unable to get the menuselect to work, it said to install ncurses but its already installed, ill pb it one moment |
| 20:45.17 | Qwell | No, stop. You are wasting your time. |
| 20:45.19 | [TK]D-Fender | voipeng, ncurses-devel |
| 20:45.21 | Qwell | Zaptel will never work with that kernel. Period. |
| 20:45.22 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-ieeregorzcykngua) |
| 20:45.23 | voipeng | oh ok |
| 20:45.45 | leifmadsen | reboot to an older kernel |
| 20:45.52 | leifmadsen | you ran yum update and are using the latest centos install |
| 20:45.57 | leifmadsen | you need to use centos 5.4 |
| 20:46.03 | Qwell | leifmadsen: It's a new server. |
| 20:46.10 | leifmadsen | I know |
| 20:46.11 | [TK]D-Fender | And an ancient kernel |
| 20:46.13 | leifmadsen | that's the problem |
| 20:46.17 | [TK]D-Fender | modules/2.6.18-274.12.1.el5PAE |
| 20:46.19 | Qwell | [TK]D-Fender: it's newer than Zaptel. |
| 20:46.20 | leifmadsen | he's using centos 5.7 now |
| 20:46.24 | [TK]D-Fender | Zaptel should work fine on that.. |
| 20:47.40 | leifmadsen | voipeng: anyways, for your particular problem I believe you want to try with centos 5.4 and then things should just install fine |
| 20:48.04 | voipeng | hmm ok wouldnt i have to rebuild it then? |
| 20:48.21 | leifmadsen | yes |
| 20:48.24 | leifmadsen | maybe |
| 20:48.36 | leifmadsen | you could try rebooting back into an older kernel which shouldbe available via grub |
| 20:48.54 | voipeng | ok ill have to wait after hours, they started forwarding phones from the server |
| 20:48.56 | voipeng | thank you |
| 20:49.40 | [TK]D-Fender | voipeng, I would start working on that DAHDI migration I recommended months ago/ |
| 20:50.54 | MindTheGap | sorry, had to leave fo a while. anyway, _Corey_ , any other thing comes thtough your mind regarding management of queues? |
| 20:51.55 | _Corey_ | MindTheGap: I could recommend a couple commercial solutions privately |
| 20:54.51 | Netgeeks | what is the md5 secret hash format in 1.8? <username>:<realm<:<secret> hashed? |
| 20:54.58 | leifmadsen | yes |
| 20:55.24 | Netgeeks | *boggle* hrm, I must be doing something wrong |
| 20:55.48 | leifmadsen | echo "lmadsen:my_realm:welcome" | md5sum |
| 20:55.52 | leifmadsen | shold be it I think |
| 20:55.55 | Qwell | nope! |
| 20:55.56 | Netgeeks | -n |
| 20:55.57 | Qwell | echo -n |
| 20:56.08 | leifmadsen | you were just waiting for me to forget seomthing |
| 20:56.14 | Qwell | all the time |
| 20:56.28 | Netgeeks | but thats okay, I'm using a python script to build the md5sum, and I verified that it produces the same thing as the echo command |
| 20:56.31 | Netgeeks | I see my error... |
| 20:57.05 | Netgeeks | yep.... |
| 20:57.18 | Netgeeks | ldap ldif files do not like extra spaces at the end of a line |
| 20:57.26 | jaytee | wow, CentOS 5.4? where could you get the iso for that? Last time I downloaded CentOS it was at 5.7 and I couldn't find any earlier versions on any of the mirror sites |
| 20:58.06 | Netgeeks | i have an archive I keep of all linux iso images that I download... I'm sure I have a centos 5.4 in there |
| 20:58.14 | Netgeeks | heck, I have a fedora core 2 iso |
| 20:58.28 | *** join/#asterisk Alborracho (c819d8de@gateway/web/freenode/ip.200.25.216.222) |
| 20:58.54 | Alborracho | Hi everybody |
| 20:59.51 | Alborracho | Can anyone help with a problem with a digium card, loks like asterisk cant comunicate with the card, i don see anything with "ss7 dump" and dahdi_tool shows all E1s in green without errors |
| 21:00.55 | Alborracho | im using asterisk 1.4 and chan_ss7-2.1 |
| 21:08.35 | akrohn | drivers? message logs? |
| 21:10.21 | *** join/#asterisk picard276 (~chatzilla@ip68-111-86-140.oc.oc.cox.net) |
| 21:10.31 | picard276 | hey... is there anyway to use USSD over VoIP |
| 21:10.45 | Qwell | What is USSD? But, no. |
| 21:10.47 | Alborracho | picard276: ussd over sigtran |
| 21:10.58 | Alborracho | but it is not voip |
| 21:11.19 | picard276 | what is sigtran? |
| 21:11.44 | picard276 | connection to ss7 server |
| 21:12.15 | picard276 | http://www.projectdiastar.org/ |
| 21:12.20 | jkroon | Alborracho, what prevents it? it's a call to a number, send some text bi-directionally, so why not? |
| 21:12.21 | picard276 | but i have more of a question of how that technology works? |
| 21:12.44 | picard276 | if i initiate a USSD request on my moblie device... pointed to my DID |
| 21:13.04 | picard276 | it will obviously not go through... but why not? is there a driver or tool in order to then accept that? or does it have to come from a PSTN line? |
| 21:13.06 | Alborracho | akrohn: in /varlog/messages got only clear alarms from yellow, asterisk messages show ERROR[4299] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory |
| 21:13.27 | picard276 | is it depedant on my DID carrier? |
| 21:13.32 | picard276 | im confused on where it breaks down |
| 21:14.38 | jaytee | ah! found a 5.4 iso in Alaska at the Arctic Region Supercomputing Center mirror |
| 21:16.10 | picard276 | Alborracho do u knw where it breaks down |
| 21:16.19 | picard276 | as in.. can a USSD request be sent over a DID and then some driver pick it up? |
| 21:16.32 | picard276 | or some piece of hardware ... or do i need a special connection like a PSTN connection of some sort? |
| 21:17.19 | Alborracho | picard276: i've only work with ussd over SS7 (E1s) and SIGTRAN (ETHERNET) but never heard of USSD in VoIP |
| 21:17.54 | picard276 | http://www.projectdiastar.org/ |
| 21:18.33 | picard276 | Alborracho but im confused on what is the SIGTRAN gateway... do i need to be hooked into my mobile network.. do i need a special Phone NUmber.. etc... or can i setup over the internet a sigtran conncetion hooked into a phone number.. im confused on the logistics |
| 21:18.41 | picard276 | is it like SIP |
| 21:18.53 | picard276 | but for mobile operators... (confused on the actuality of what is) |
| 21:19.16 | Alborracho | picard276: you need to be connected to a mobile telco, and more specific to the HLR |
| 21:19.28 | Alborracho | USSD is part of the GSM protocol |
| 21:19.38 | picard276 | right |
| 21:19.40 | picard276 | i understand that.. |
| 21:19.56 | picard276 | but my question is (example) if i am "roaming" on ATT (i do not own the ATT network) |
| 21:20.18 | picard276 | and i send a USSD MO command (mobile originated) so like *171#phonenumber (my phone number on my server) |
| 21:20.30 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 21:20.40 | picard276 | Att will route that to my phone number ... but does that phone number have to be connected somehow.. it cannot be a VOIP did? |
| 21:21.16 | Alborracho | if att donw have the *171# configurated it will carry your request to your telco |
| 21:21.40 | picard276 | If |
| 21:21.51 | picard276 | so if I created something *172312312123123123# or something ridiculously long |
| 21:22.07 | picard276 | no telco would have that configued.. and they would deliver the message to #number? |
| 21:22.35 | picard276 | but that #number.. has to be ... on a certain type of line? or can it be a Voip based DID (you can see my understand is breaking down here) |
| 21:23.11 | Alborracho | no, imagine that *171#............ and everything before that belongs to the same route, in the end of that ropute should be a ussd gateway to get the request |
| 21:23.29 | Alborracho | but a phone, i dont think so |
| 21:23.41 | Alborracho | maybe its posible, but i never tried thatr |
| 21:23.57 | picard276 | at the end of the route.. |
| 21:24.14 | picard276 | right... which could be initiated over a sigtran connection.. |
| 21:24.23 | picard276 | the part that i am not understand is what does the mobile operator "control" |
| 21:24.31 | picard276 | i know the feature code *123124#number |
| 21:25.02 | picard276 | i thought it was universal that *(featurecode)#(phonenumber).... in a sense the mobile operator will deliver automatically a feature code of 121 to the phone number #number assuming the #number has a USSD gateway programmed to it |
| 21:25.23 | Alborracho | nop |
| 21:25.26 | picard276 | but you cannot send a *featurecode#number unless you are over GSM (what about of sigtran?) ... and do i need a special type of phone |
| 21:25.32 | Alborracho | it can be anything |
| 21:25.40 | picard276 | what can be anything? |
| 21:25.50 | carrar | Q |
| 21:25.58 | Alborracho | we sell mobile banking via ussd, ringtones and that kind of thing |
| 21:26.11 | picard276 | ok |
| 21:26.14 | Alborracho | anything |
| 21:26.15 | Alborracho | we sell sms packets via ussd |
| 21:26.21 | picard276 | ok |
| 21:26.36 | picard276 | so how does the user access your service? |
| 21:26.50 | Alborracho | *123# and we print a menu |
| 21:27.00 | picard276 | but on which network? |
| 21:27.09 | Alborracho | then the user navigates like ins the sim card but faster |
| 21:27.33 | Alborracho | telcos network |
| 21:27.34 | picard276 | is *123# you have to go to each carrier and be like.. hey make *123# point to this USSD gateway |
| 21:27.57 | Alborracho | yeah, we have a box inside telco |
| 21:28.04 | picard276 | for each telco though |
| 21:28.08 | Alborracho | conected to the HLR |
| 21:28.11 | Alborracho | yup |
| 21:28.13 | picard276 | so if u want to work in each network.. |
| 21:28.19 | picard276 | so you have a separate box for ATT and Tmobile |
| 21:28.49 | Alborracho | yes, ussd requieres low latency |
| 21:28.57 | picard276 | but is that a latency issue |
| 21:29.00 | picard276 | or is it that |
| 21:29.01 | Alborracho | i dont know if you can work with a box outside |
| 21:29.09 | picard276 | if i were to type *123# on tmobile |
| 21:29.16 | picard276 | it would not get routed to your box.. tmobile has no idea where to route that? |
| 21:29.25 | picard276 | or since you are in ONE HLR you can then be routed anywhere in the world? |
| 21:30.12 | Alborracho | exactly if you are tmobile user and type 123, they don know what it means |
| 21:30.14 | *** join/#asterisk twanny796 (~twanny@78.133.48.50) |
| 21:30.31 | Alborracho | but if you are att roaming in tmobile and type 123 they will carry that to your telco |
| 21:31.01 | picard276 | i c ic |
| 21:31.36 | picard276 | so u need the box in HLR of the subscriber you want to attach to |
| 21:31.54 | picard276 | so a box in ATT... but you can roam anywhere.. as long as u have ur ATT sim in and it will work.. |
| 21:32.04 | Alborracho | yes, at least thats how we work, i dont know if there are other ways to do that |
| 21:32.16 | picard276 | but... if u have a tmobile sim.. even if ur on ATT network (roaming) or any other network... then it will not work |
| 21:32.18 | picard276 | gotchya |
| 21:32.20 | picard276 | that makes sense.. |
| 21:32.26 | picard276 | becuase i was not sure how the USSD call gets routed.. |
| 21:32.32 | Alborracho | yes |
| 21:34.36 | picard276 | so even if u typed like *18005557788 |
| 21:34.41 | picard276 | it would not send the *to the 1800 number? |
| 21:37.16 | picard276 | well different question i guess. |
| 21:37.26 | picard276 | is there anyway to capture more information on an incoming call |
| 21:37.30 | picard276 | i know i can get CallerID etc.. |
| 21:37.42 | picard276 | but is there anyway to get more information than just that from a phone call? |
| 21:38.31 | WIMPy | USSDs are not related to phone numbers. |
| 21:38.48 | WIMPy | And AFAIK USSD is always routed to the HLR. |
| 21:39.05 | *** join/#asterisk edge (~IceChat7@97-64-216-2.client.mchsi.com) |
| 21:39.12 | picard276 | can you explain WIMPy? |
| 21:39.16 | Alborracho | picard276: it wont |
| 21:39.24 | picard276 | ahh sorry |
| 21:39.26 | picard276 | misread |
| 21:39.26 | WIMPy | What? |
| 21:39.27 | picard276 | always routed to HLR |
| 21:39.42 | picard276 | what about a server initiated USSD request |
| 21:39.50 | picard276 | so i send a USSD message to my phone number? |
| 21:40.01 | Alborracho | picard276: its fomr ussd gateway box, to hlr, to subscriber |
| 21:40.03 | edge | What is the best way to take a call and transfer it to somebody's voicemail? |
| 21:40.17 | picard276 | edge *exten |
| 21:40.21 | picard276 | exten* |
| 21:40.27 | picard276 | forget which way you do it |
| 21:40.39 | picard276 | its *exten or exten* |
| 21:40.54 | picard276 | right Alborracho .. |
| 21:40.56 | picard276 | but |
| 21:41.04 | picard276 | do i need to have sometihng in the HLR to do that? |
| 21:41.08 | picard276 | so i have my cellphone sitting here |
| 21:41.17 | picard276 | and lets theoretically say i have a USSD box here too |
| 21:41.30 | picard276 | connected to what? (internet, Sip provider, ??) |
| 21:41.31 | WIMPy | You need a connection to the Operator. |
| 21:41.46 | picard276 | either way you are saying WIMPy wether i send it or i am receiving |
| 21:41.49 | edge | picard276 i think my SPA502G phones are removing digits after the * |
| 21:42.08 | picard276 | SPA502G are those cisco/linksys ... usually you have to config dial plan differently |
| 21:42.12 | WIMPy | Direction doesn't matter, no. |
| 21:42.37 | picard276 | edge do u have the feature code enabled?" |
| 21:43.27 | picard276 | ok |
| 21:43.38 | picard276 | one last question.. Alborracho u might be able to explain this as well |
| 21:43.49 | picard276 | with WAP gateways.. i am assuming that has to be mobile side as well |
| 21:43.54 | edge | picard276 I think it might be |
| 21:44.07 | picard276 | so as an example |
| 21:44.22 | picard276 | i create a WAP gateway... and put the settings in my phone to connect to that gateway |
| 21:44.56 | picard276 | it will not connect... because my wap gateway has to be tied in at the HLR or somewere in the Mobile operator? |
| 21:45.55 | WIMPy | Yes, you again need a link to the operator. |
| 21:46.06 | picard276 | gotchya |
| 21:50.22 | edge | Does anybody have the Cisco SPA502g and know how to disable it intercepting the *xxx ? |
| 21:50.57 | WIMPy | Check (or just remove) the dialplan. |
| 21:51.29 | picard276 | edge i do not have that phone... but i would add the *xxx to the beginning of the dial plan |
| 21:53.00 | edge | picard276 i put it in the front of the phone's dialplan (and it is in asterisk's dial plan too) but it still cuts off the last digit. so if i dial *121 , it tries *12 |
| 21:53.26 | picard276 | you put it at the very front? |
| 21:55.14 | edge | picard276 it is |
| 21:55.20 | picard276 | hmmm |
| 21:55.22 | picard276 | try putting *121 |
| 21:55.25 | picard276 | at the very front.. |
| 21:55.29 | picard276 | then test that extension |
| 21:55.36 | picard276 | and how are u editing the dial plan |
| 21:55.46 | picard276 | end point manager? |
| 21:55.54 | picard276 | make sure its updating correctly too |
| 21:56.25 | edge | picard276 each phone has a web interface for configuration. I updated it there. |
| 21:56.34 | picard276 | gotchya |
| 21:56.38 | picard276 | yea |
| 21:56.47 | picard276 | try doing a manual *121 and see if that worked for exten 121 |
| 21:57.06 | edge | picard276 I'll know in a moment, it has to restart after every single change |
| 21:57.13 | picard276 | k |
| 21:57.49 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:58.16 | WIMPy | remembers those restart orgies :-( |
| 21:58.40 | WIMPy | Nice to hear that this hasn;t changed :-( |
| 21:58.42 | edge | picard276 still only accepts two digits after a * |
| 22:08.06 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 22:18.07 | Alborracho | does anyone know how to capture ss7 signaling from my digium card?, "ss7 dump" doesnt print anything |
| 22:18.36 | *** join/#asterisk shido6_ (~shido6@nat/yahoo/x-msinmkyranfjogan) |
| 22:25.54 | carrar | digium having DNS issues? |
| 22:26.19 | WIMPy | Not for me |
| 22:26.20 | k-man | anyone managed to get a cisco 7942g to work with asterisk? I'm having trouble working out the configuration files and what to put in them on my tftp server |
| 22:26.40 | carrar | specifically switchvox |
| 22:26.42 | carrar | www.switchvox.com |
| 22:26.42 | k-man | in fact I can't even get the phone to provision - haven't even go to trying to get it to work with asterisk yet |
| 22:27.15 | fenrus | k-man, check the logs on the tftp-server to see what files it tries to get |
| 22:27.26 | WIMPy | carrar: AAlso ok with a little delay. |
| 22:27.30 | fenrus | k-man, and make sure that you give it tftp-server address via dhcp-options |
| 22:27.35 | carrar | hrmm |
| 22:27.44 | carrar | I'll clear my dns servers cache |
| 22:27.59 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 22:29.41 | carrar | now they are workin |
| 22:29.44 | carrar | weird |
| 22:30.10 | carrar | Was querying their NAMERESOLVE.COM hosting company servers too |
| 22:31.22 | k-man | fenrus, yeah, I see them: http://pastie.org/3168969 but what should I put in those files? I already set up a XMLDefault.cnf.xml |
| 22:33.18 | *** join/#asterisk jeffspeff (~jeffspeff@173-11-144-149-houston.txt.hfc.comcastbusiness.net) |
| 22:33.19 | fenrus | nice |
| 22:33.38 | fenrus | theres example files for the xml syntax |
| 22:33.57 | fenrus | they should contain the configuration the phone needs, like username/password/asterisk ip |
| 22:34.00 | fenrus | etc |
| 22:35.26 | jeffspeff | I'm trying to alter the name of the CID for inbound calls... however, this isn't working. I have this placed in the step before it dials the internal phone. exten=s,n,Set(CALLERID="Artius Inbound" <${CALLERID(num)}>) it prints properly on the cli output, but the phone doesn't show this modded CID. |
| 22:35.57 | [TK]D-Fender | jeffspeff: Show us |
| 22:36.28 | jeffspeff | just a sec |
| 22:36.31 | WIMPy | you didn't specify what part of the CALLERID you want to change. The pareameter is missing. |
| 22:37.03 | jeffspeff | WIMPy, i'm wanting it to change the name but leave the number |
| 22:37.26 | [TK]D-Fender | jeffspeff: You aren't setting the callerID there |
| 22:37.45 | WIMPy | Set CLLERID(name). |
| 22:37.48 | [TK]D-Fender | jeffspeff: You are setting some random variable. |
| 22:38.16 | jeffspeff | but if i just specify (name) then that'll remove the number completely won't it? |
| 22:38.29 | WIMPy | No |
| 22:38.31 | [TK]D-Fender | that should be (all) based on what you're setting it to |
| 22:38.33 | WIMPy | What makes you think so? |
| 22:38.57 | [TK]D-Fender | WIMPy: exten=s,n,Set(CALLERID="Artius Inbound" <${CALLERID(num)}>) <- look at the format |
| 22:39.27 | [TK]D-Fender | Now of course since he's leaving the number alone effectively he should be just setting the name |
| 22:39.32 | *** join/#asterisk shido6_ (~shido6@nat/yahoo/x-stpghwybldxqbugm) |
| 22:39.33 | WIMPy | Oh, misread that as two parts. |
| 22:39.38 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 22:39.45 | WIMPy | yes |
| 22:41.16 | p3nguin | You either Set(CALLERID(all)=NAME WITHOUT QUOTES <number>) or just Set(CALLERID(name)=NAME WITHOUT QUOTES). |
| 22:41.58 | p3nguin | In your case, Set(CALLERID(name)=Artius Inbound) would be the correct thing to do. |
| 22:42.48 | jeffspeff | thanks everybody, as always, you all were correct. :) i don't know where i got the idea that if you set one and not the other it removes the other... |
| 22:42.55 | WIMPy | It might be neccessary to also set CALLERID(name-pres)=allowed. |
| 22:43.12 | *** part/#asterisk wesphillips (~wphill04@192.160.117.129) |
| 22:43.56 | p3nguin | Set(CALLERID(name)=Artius Inbound) <------ |
| 22:45.18 | jeffspeff | p3nguin, yes, that worked |
| 22:46.00 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ylhprtzzbawpiatk) |
| 22:53.11 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 22:53.21 | *** join/#asterisk bakermd (~bakermd@38.104.0.142) |
| 22:54.29 | Alborracho | anyone knows how to capture ss7 signaling besides "ss7 dump" i think that asterisk cannont comunicate with my digium card |
| 22:55.43 | *** join/#asterisk serafie (~erin@nat/digium/x-wxxglboitnnympmh) |
| 22:56.27 | *** part/#asterisk bakermd (~bakermd@38.104.0.142) |
| 23:10.46 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-snwcjansjyhhqnst) |
| 23:17.25 | *** join/#asterisk libryder (~david@209.33.214.243) |
| 23:17.57 | libryder | during an active call, can i set an arbitrary channel variable using the SET application? |
| 23:21.31 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-rudoyslykfgvtjvz) |
| 23:21.45 | libryder | i may have just found my answer |
| 23:22.47 | *** join/#asterisk wdoekes2 (~walter@wjd.osso.nl) |
| 23:24.54 | *** join/#asterisk skrull (~skrull@200-203-68-83.smace701.dsl.brasiltelecom.net.br) |
| 23:26.34 | ChannelZ | 42 |
| 23:31.41 | picard276 | hey guys.. |
| 23:31.58 | picard276 | is there a way to grab data from an incoming call |
| 23:32.05 | picard276 | more than what is currently displayed? |
| 23:33.33 | libryder | 42 was close |
| 23:36.20 | skrull | Hi. I have a working setup with asterisk and spa3102. I´m getting ¨Auto fallthrough, channel 'SIP/pstn1-0000000f' status is 'UNKNOWN'¨ when I dial local operator call-center. If I shutdown asterisk, spa3102 handles the call normally. Any thoughts on this? |
| 23:38.06 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 23:38.16 | joobie | hey guys.. anyone know with the SPA 508G how the 8 lines are useD? |
| 23:38.37 | joobie | i mean,can you use part of them as presence quickdial extensions? |
| 23:54.26 | picard276 | hey |
| 23:54.29 | picard276 | Alborracho u still there? |